Protection of speaker from excess excursion

ABSTRACT

Systems and methods for protecting a loudspeaker from excessive excursion include an audio source, an adaptive excursion protection filter, an audio clipper, an inverse excursion protection filter, an amplifier and a loudspeaker. The system performs operations including receiving an audio signal, applying an excursion protection filter, the excursion protection filter adapting in real-time to one or more speaker conditions, clipping the audio signal, applying an inverse excursion protection filter, and amplifying, using an amplification circuit, the audio signal for output to the speaker.

TECHNICAL FIELD

Various embodiments of the present disclosure relate generally to thereduction of speaker distortion and/or the protection of loudspeakersduring operation. More particularly, for example, the present disclosurerelates to systems and methods for protecting loudspeakers from excessexcursion.

BACKGROUND

Modern consumer electronic devices, such as mobile phones and tabletcomputers, are typically designed to meet various small form factor, lowpower and low-cost goals. The components used in these devices,including loudspeakers, may be low-cost components designed to fit intosmall spaces. A drawback of these smaller loudspeakers is a reducedacoustic output that can be delivered from the device to the user. Tosatisfy a user's desire for more loudness or volume delivered throughthe loudspeaker, the loudspeaker may be driven close to its mechanicallimits, which can potentially lead to catastrophic loudspeaker failuresif the limits are exceeded. In view of the foregoing, there is acontinued need for systems and methods that improve acoustic performanceof loudspeaker(s) in consumer electronic devices while protecting theloudspeaker(s) from damage.

SUMMARY

Systems and methods are disclosed herein for protecting a loudspeakerand/or reducing distortion due to excess excursion. In variousembodiments, systems and methods for protecting a loudspeaker fromexcessive excursion include an audio source, an adaptive excursionprotection filter, an audio clipper, an inverse excursion protectionfilter, an amplifier and a loudspeaker. The system performs operationsincluding receiving an audio signal, applying an excursion protectionfilter, the excursion protection filter adapting in real-time to one ormore speaker conditions, clipping the audio signal, applying an inverseexcursion protection filter, and amplifying, using an amplificationcircuit, the audio signal for output to the speaker.

In some embodiments, the system includes a logic device furtherconfigured to perform operations comprising estimating filtercoefficients for the excursion protection filter and the inverseexcursion protection filter based at least in part on the amplifiedaudio signal. The operation of estimating the filter coefficients mayfurther comprise receiving current and voltage data from the amplifier.The system may further include a cone measurement sensor configured tomeasure speaker excursion of the output audio signal, wherein the one ormore speaker conditions comprises a speaker excursion measurement. Thelogic device may be further configured to perform operations comprisingestimating a threshold value and applying the threshold value whenclipping the audio signal.

In some embodiments, the operation of estimating the excursion filterfurther comprises estimating a complex impedance of the speaker andestimating a direct current resistance. The operation of estimating theexcursion filter may further comprise computing a motion sensitivityfunction representing motion of the cone as a function of appliedvoltage and estimating two or more peaks of the computed motionsensitivity function. The operation of estimating the excursion filtermay further comprise imposing a slew-rate limit on the peak frequenciesand amplitudes and creating a model infinite impulse response filterbased at least in part on the estimated peaks.

In some embodiments, the logic device is further configured to performoperations comprising testing parameter values for a current frame ofthe audio signal, the parameter values comprising one or more parametersused in the estimation of the excursion filter and identifying the frameas invalid if a parameter is detected outside a predetermined range. Thelogic device may be further configured to perform operations comprisingtuning the excursion protection filter through a process comprisingmeasuring a scale factor, determining a clipping level threshold value,and verifying the clipping level threshold value.

The scope of the disclosure is defined by the claims, which areincorporated into this section by reference. A more completeunderstanding of embodiments of the present disclosure will be affordedto those skilled in the art, as well as a realization of additionaladvantages thereof, by a consideration of the following detaileddescription of one or more embodiments. Reference will be made to theappended sheets of drawings that will first be described briefly.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a flow diagram illustrating real-time speaker protection, inaccordance with one or more embodiments.

FIG. 2 is a flow diagram illustrating a model estimation for real-timespeaker protection, in accordance with one or more embodiments.

FIG. 3 illustrates a real-time adaptive algorithm for updating the modelestimation filter, in accordance with one or more embodiments.

FIG. 4 illustrates an offline tuning algorithm for tuning a modelestimation filter, in accordance with one or more embodiments.

FIG. 5 illustrates an example processing system configured to protect aspeaker from excess excursion, in accordance with one or moreembodiments.

Embodiments of the present disclosure and their advantages are bestunderstood by referring to the detailed description that follows. Itshould be appreciated that like reference numerals are used to identifylike elements illustrated in one or more of the figures, whereinshowings therein are for purposes of illustrating embodiments of thepresent disclosure and not for purposes of limiting the same.

DETAILED DESCRIPTION

The present disclosure provides systems and methods for protecting aloudspeaker from excess excursion during operation. In variousembodiments, a louder sound is obtained from a loudspeaker by drivingthe loudspeaker with higher voltage/power than the standardspecification limits for the loudspeaker. Systems and methods areconfigured to protect the loudspeaker from excessive excursion of thecone of the speaker, which can cause distorted sound and/or damage tothe loudspeaker.

An embodiment of the present disclosure will be described with referenceto FIG. 1, which illustrates an example real-time algorithm data flowfor speaker protection. The algorithm 100 includes a feedforward path102 that receives audio data from an audio source 104 and processes theaudio data for output through a loudspeaker 114. The feed-forward path102 may operate at a sample rate suitable for high quality audioplayback, such as 48 KHz. The other paths (e.g., path 120 to 108; path112 to 130; 130 to 106; and 130 to 110) in the algorithm 100 areconfigured to operate at a lower rate, such as a frame rate of 100 Hz.In operation, the excursion of the speaker 114 is limited by the valueof the clipping threshold (thr) 120, which is used in the look-aheadclipper 108. The look-ahead clipper 108 is configured to avoid the highfrequency distortion that would be created by using a simpleinstantaneous threshold.

The algorithm 100 creates a model that predicts the cone excursion ofthe loudspeaker given a drive voltage waveform and uses this model tolimit the distortion to a safe value. The algorithm 100 operates inreal-time to apply this limiting operation and optionally dynamicallyupdates parameters of the model based on the real-time current andvoltage applied to the speaker.

An offline tuning algorithm measures the behavior of the loudspeaker 114and generates an appropriate initial model and other parameters for thereal-time algorithm 100.

The approximate model is configured to operate in real time to predictthe excursion of the loudspeaker 114 from the applied voltage waveform.The approximate model may be implemented as an infinite impulse response(IIR) filter S 106 specified by SOS coefficients (e.g., coefficients insecond-order sections of IIR filter). The filter S applied in block 106has a stable inverse filter S⁻¹, which is applied in block 110. Theloudspeaker excursion is limited to a safe value by applying theapproximate model filter 106 (e.g., IIR filter S) followed by alook-ahead clipper 108 or other limiter followed by the inverse filter110 (e.g., inverse filter S⁻¹). The excursion model S is continuallyupdated in real-time to adapt to changes in properties of theloudspeaker due to factors such as aging and temperature.

An initial excursion model 130 is estimated for the loudspeaker 114. Forexample, for a given loudspeaker the initial excursion model 130 isestimated based on a method combining electro-mechanical loudspeakertheory and heuristics enabling an approximate model to be calculatedwith low computational power. This procedure may use the current (I) andvoltage (V) data from the amplifier 112 (e.g., a class-D amplifier) thatdrives the loudspeaker 114 and may optionally use a sensor 132 formeasuring instantaneous cone excursion, e.g., such as with a laserdisplacement measuring device.

In contrast to conventional algorithms, the algorithm 100 is configuredto track more than one peak in the excursion frequency response. Mostmini-speakers have more than one peak, for example, because of thecommon use of ports, passive radiators, or multi-motors to addlow-frequency response.

Referring to FIG. 2, an embodiment for estimating the model will now bedescribed. In the illustrated embodiment, the model S is estimated in230 based on the time-domain current I and voltage V (e.g., IV 202) atthe speaker terminal. This data can be obtained, for example, throughcircuitry integrated with the class-D amplifier 112 of FIG. 1. The inputdata IV is provided to a subband analysis block 210, which is configuredto convert the time domain input signals to frequency domain subbands,for example, by a short-term Fourier transform. An adaptive filter 220is configured to estimate the speaker complex impedance Z(f) in thefrequency domain. In various embodiments, the absolute value of thespeaker impedance |Z(f)| will show a prominent peak at the mechanicalresonant peak of the loudspeaker cone and its elastic constraints (e.g.,a combination of the effect of the spider, surround, and air reaction).This is due to the back electric and magnetic fields (EMF) induced bythe motion of the speaker/voice coil in its magnetic field. When themotion velocity is higher, such as at resonance, the back EMF opposesthe drive voltage and the resulting smaller effective drive voltageapplied to the DC resistance of the coil results in less current.

The voltage in the frequency domain, V(f), is related to the current,I(f), velocity of speaker coil and cone, velocity(f), DC resistance ofthe voice coil, Rdc, and a force constant, BL, which is the product ofthe (average) magnetic field B and the length L of the voice coilwinding, as follows:V(f)=Rdc*I(f)+velocity(f)*Bl

Referring to FIG. 3, embodiments of a real-time adaptive algorithm forupdating the filter S will now be described. The algorithm 300 receivesinput data representing the speaker impedance, such as frames of Z(f)data, in step 310. In some embodiments, the frames may occur, forexample, at 8 millisecond intervals, thus representing many samples ofactual audio time-domain data.

In step 312, the algorithm estimates the complex impedance of thespeaker Z(F) using a least mean squares (LMS) or other algorithm toobtain the complex impedance Z(f) such that norm(V(f)−Z(f)*I(f)) isminimized. The norm function can be the squared magnitude summed overfrequency.

In step 314, the algorithm estimates the DC resistance Rdc as theminimum of abs(Z(f)). Due to phase cancellation, the function abs(Z(f))can have values that are less than the true value of Rdc. In variousembodiments, a histogram of values of abs(Z(f)) is computed and a valueis chosen from this histogram representing a low percentile value ofabs(Z(f)). For example, a 10-percentile value may be used, which is thevalue that is lower than 90% of the values of abs(Z(f)) over allrelevant frequencies.

In step 316, the algorithm computes an effective motion sensitivityfunction Gmot(f) that represents the motion of the cone as a function ofapplied voltage: Gmot(f)=(1−Rdc/Z(f))/f. In the illustrated embodiment,the algorithm divides by f because motion is the integral of velocityover time, and the duration of each cycle of the signal components at fis inversely proportional to f. Thus, motion tends to be larger at lowerfrequencies.

In step 318, the algorithm estimates two (or more) peaks of |Gmot(f)|.In some embodiments, the algorithm picks peaks in the coarse grid offrequencies f used in the frequency-domain representation and thenrefines the estimate of peak frequency and amplitude by quadraticinterpolation of values of |Gmot(f)|. For many loudspeakers such as themini-speakers used in PC's, phones, and smart devices, the algorithmwill pick two peaks: (i) a peak corresponding to the resonant frequencyof the speaker; and (ii) a lower frequency peak that is the combinedresult of the 1/f factor in Gmot together with a possible low-frequencyacoustic resonance produced by a port in the speaker enclosure.

In step 320, the algorithm imposes a slew-rate limit on the peakfrequencies and amplitudes to prevent the filters being created (S andS⁻¹) from changing too rapidly, which can result in audible artifacts.In other words, we do not immediately use the observed peaks andamplitudes—we maintain actionable values of peaks and amplitudes whichmove toward the observed peaks and amplitudes at no more than a certainconstant rate.

In step 322, the algorithm creates a model IIR filter S that matches theactionable peaks. The filter may be generated with two (or more) pairsof poles and zeros, using a filter design algorithm, and may berepresented by SOS coefficients.

In step 324, the stable inverse S⁻¹ filter is determined (e.g., as usedin the speaker protection topology of FIG. 1). This inverse is obtainedby simply exchanging the numerator and denominator terms in the SOSrepresentation, followed optionally by normalization of thezeroth-numerator coefficient to 1.0.

In step 326, the algorithm verifies the validity of the input data frameby running tests to verify that the various parameters from steps312-324 are within a normal range. If any of these tests fails, theframe is marked as “invalid” and no modification is made in theactionable values of peaks and amplitudes.

Referring to FIG. 4, an offline “tuning” algorithm 400 will now bedescribed in accordance with one or more embodiments. In step 410, thealgorithm estimates an initial model filter S. In the illustratedembodiment, a broadband signal at a sample rate of 48 KHz is sent to thespeaker. This broadband signal contains energy at a range of frequenciesbelow Nyquist (e.g., all frequencies below Nyquist) and has enoughenergy to create excursion on the loudspeaker to a sufficient degree tobe able to measure the electrical impedance of the loudspeaker voicecoil accurately, but not so much energy that significant heating of thevoice coil or other damage occurs. In one implementation, the signalincludes a series of short bursts of pink noise at approximately −12 dBrelative to maximum RMS voltage drive. The instantaneous current andvoltage may be measured by class-D amplifier with voltage/current sensecircuitry. These time-domain signals are converted to frequency-domainby a suitable short-term Fourier transform in a subband analysisprocess. In one implementation, for example, the 48 KHz data isdown-sampled to 16 KHz and overlapping frames of 320 16 KHz samples areconverted to 64 complex frequency-domain values current I(f) and voltageV(f). Steps 312 through 322 from FIG. 3 may be performed to estimate aninitial filter S.

In step 412, a scale factor is measured. In one implementation, a seriesof increasing amplitude short sine wave bursts at the frequency of thedominant peak of S are sent to the loudspeaker after passing through thepath 102 of FIG. 1, with a nominal constant value of clipping thresholdthr 120. The excursion of the loudspeaker 114 is measured by a laserranging device or other cone measurement sensor 132. The amplitude ofthe component of the excursion at the frequency of the bursts iscalculated by a matched filter technique (for example, the rms value ofa Hann-windowed sine convolved with the measured excursion waveform).The excursion amplitude will asymptotically reach a maximum at thenominal threshold thr times the scale factor. The scale factor can thenbe calculated as the asymptotic maximum divided by the nominal thresholdthr. This scale factor is then incorporated in the values of the SOScoefficients representing filter S.

In step 414, a clipping level threshold thr is established for onlineuse. To apply the algorithms disclosed herein in real time, a clippingthreshold thr is determine that will provide the highest possible soundlevel without excessive distortion. To determine thr, a series of sinebursts are sent to the speaker as illustrated in FIG. 1, with a nominalvalue of thr which is initially set too high (for example a thr of 1.0mm for a typical mini-speaker for which an actual safe thr is known tobe about 0.5 mm.), at a number of frequencies (e.g., from 100 Hz to 8000Hz in 1/12 octave logarithmic spacing), and increasing amplitudes (e.g.,−12 dB FS to 0 dB FS in 3 dB steps). Harmonic distortion of theexcursion signal (measured by laser or other sensing component 132) iscalculated for each burst. A threshold thr is set such that if theexcursion had been limited to that value, the harmonic distortion wouldnot exceed a pre-specified limit, for example 10% (corresponding to 20dB SDR).

In step 416, the threshold value thr is verified. In one implementation,a series of sine bursts at full scale is sent to the speaker (e.g., viathe topology of FIG. 1) with the value of thr calculated in step 414 andscale factor calculated in step 412. The excursion of the speaker ismeasured by laser or other sensor 132 and plotted as a function of burstfrequency. In various embodiments, the excursion is limited at everyfrequency to a value close to thr.

In some embodiments, it may be desirable to run the real-time algorithmnon-adaptively. In this case the filter S, threshold thr, and scalefactor can be determined by the tuning algorithm. S is then keptconstant for live use. This eliminates the adaptation steps, simplifyingthe implementation and reducing the computational requirements. Thenon-adaptive algorithm could be used, for example, to control distortionin a system on a chip implementation for embedded applications.

As discussed, the various techniques provided herein may be implementedby one or more systems which may include, in some embodiments, one ormore subsystems and related components thereof. For example, FIG. 5illustrates a block diagram of an example processing system 500 inaccordance with an embodiment of the disclosure. The system 500 may beused to implement any desired combination of the various components,circuits, processing steps, and other operations described herein.Although a variety of components are illustrated in FIG. 5, componentsmay be added and/or omitted for different types of devices asappropriate in various embodiments.

As shown, processing system 500 includes an audio processing system 510which includes a memory 520, a processor 540, and audio output circuitry550. Processor 540 may be implemented as one or more logic devices suchas a microprocessor, microcontroller, application specific integratedcircuit (ASIC), programmable logic device (PLD) (e.g., fieldprogrammable gate array (FPGA), complex programmable logic device(CPLD), system on a chip, or other types of programmable devices.

In some embodiments, processor 540 executes machine readableinstructions (e.g., software, firmware, or other instructions) stored inmemory 520. In this regard, memory 520 may include logic to cause theprocessor 540 to perform any of the various operations, processes, andtechniques described herein. In other embodiments, processor 540 and/ormemory 520 may be replaced and/or supplemented with dedicated hardwarecomponents to perform any desired combination of the various techniquesdescribed herein.

Memory 520 may be implemented as a machine-readable medium storingvarious machine-readable instructions and data. For example, in someembodiments, memory 520 may store one or more algorithms as machinereadable instructions that may be read and executed by processor 540 toperform the various techniques described herein, including a subbandanalysis algorithm 522, filter estimation algorithm 524, offline tuningalgorithm 526, and/or other executable logic. Memory 520 may also storedata used by the audio processing system 510. In some embodiments,memory 520 may be implemented as non-volatile memory (e.g., flashmemory, hard drive, solid state drive, or other non-transitorymachine-readable mediums), volatile memory, or combinations thereof.

The system 500 further includes audio source 570 and one or morespeakers 560. In operation the audio source 570 provides audio data forprocessing by the audio output circuitry and playback by the one or morespeakers 560. The system 500 may be used to perform processes disclosedherein to protect the one or more speakers 560 from excess excursion. Insome embodiments, each of the one or more speakers 560 has acorresponding excursion protection filter disclosed herein. The system500 may further include other speaker protection techniques includingtemperature and power protection of the speakers 560.

As used herein, the terms “substantially,” “about,” and similar termsare used as terms of approximation and not as terms of degree, and areintended to account for the inherent deviations in measured orcalculated values that would be recognized by those of ordinary skill inthe art. Further, the use of “may” when describing embodiments refers to“one or more embodiments of the present disclosure.” As used herein, theterms “use,” “using,” and “used” may be considered synonymous with theterms “utilize,” “utilizing,” and “utilized,” respectively. Also, theterm “exemplary” is intended to refer to an example or illustration.

The electronic or electric devices and/or any other relevant devices orcomponents according to embodiments of the present disclosure describedherein may be implemented utilizing any suitable hardware, firmware(e.g. an application-specific integrated circuit), software, or acombination of software, firmware, and/or hardware. For example, thevarious components of these devices may be formed on one integratedcircuit (IC) chip or on separate IC chips. Further, the variouscomponents of these devices may be implemented on a flexible printedcircuit film, a tape carrier package (TCP), a printed circuit board(PCB), or formed on one substrate. Further, the various components ofthese devices may be a process or thread, running on one or moreprocessors, in one or more computing devices, executing computer programinstructions and interacting with other system components for performingthe various functionalities described herein. The computer programinstructions are stored in a memory which may be implemented in acomputing device using a standard memory device, such as, for example, arandom-access memory (RAM). The computer program instructions may alsobe stored in other non-transitory computer readable media such as, forexample, a CD-ROM, flash drive, or the like. Also, a person of skill inthe art should recognize that the functionality of various computingdevices may be combined or integrated into a single computing device, orthe functionality of a particular computing device may be distributedacross one or more other computing devices without departing from thespirit and scope of the exemplary embodiments of the present disclosure.

Embodiments described herein are examples only. One skilled in the artmay recognize various alternative embodiments from those specificallydisclosed. Those alternative embodiments are also intended to be withinthe scope of this disclosure. As such, the embodiments are limited onlyby the following claims and their equivalents.

What is claimed is:
 1. A method for processing an audio signal foroutput through a speaker, the method comprising: receiving the audiosignal; adapting an adaptive excursion protection filter in real-time toone or more speaker conditions based at least in part on a motion of acone of the speaker as a function of applied voltage in the audiosignal; applying the adaptive excursion protection filter to the audiosignal; clipping the audio signal; applying an inverse excursionprotection filter; and amplifying, using an amplification circuit, theaudio signal to generate an output signal for output to the speaker. 2.The method of claim 1, further comprising: estimating filtercoefficients for the adaptive excursion protection filter and theinverse excursion protection filter based at least in part on theamplified audio signal.
 3. The method of claim 2, wherein estimating thefilter coefficients for the adaptive excursion protection filter and theinverse excursion protection filter further comprises: receiving currentand voltage data from the amplification circuit; converting the currentand voltage data to frequency domain subbands; and adaptively estimatinga complex impedance of the speaker.
 4. The method of claim 1, furthercomprising physically measuring speaker excursion of the output audiosignal using a sensor, wherein the one or more speaker conditionscomprises a speaker excursion measurement.
 5. The method of claim 1,further comprising estimating a threshold value and applying thethreshold value when clipping the audio signal.
 6. The method of claim2, wherein estimating the filter coefficients for the adaptive excursionprotection filter and the inverse excursion protection filter furthercomprises: estimating a complex impedance of the speaker; and estimatinga direct current resistance.
 7. The method of claim 6, whereinestimating the filter coefficients for the adaptive excursion protectionfilter and the inverse excursion protection filter further comprises:computing a motion sensitivity function representing the motion of thecone of the speaker as the function of applied voltage of the audiosignal; and estimating two or more peaks of the computed motionsensitivity function.
 8. The method of claim 7, wherein estimating thefilter coefficients for the adaptive excursion protection filter and theinverse excursion protection filter further comprises: computing themotion sensitivity function based at least in part on a measured DCresistance of a voice coil and pink noise; imposing a slew-rate limit onthe peak frequencies and amplitudes; and creating a model infiniteimpulse response filter based at least in part on the estimated peaks.9. The method of claim 6, further comprising testing parameter valuesfor a current frame of the audio signal, the parameter values comprisingone or more parameters used in the estimation of the filter coefficientsfor the adaptive excursion protection filter and the inverse excursionprotection excursion filter and identifying the frame as invalid if aparameter is detected outside a predetermined range.
 10. The method ofclaim 1, further comprising tuning the adaptive excursion protectionfilter through a process comprising: measuring a scale factor;determining a clipping level threshold value; and verifying the clippinglevel threshold value.
 11. A system comprising: an amplifier configuredto drive a speaker to output a processed audio signal; a logic deviceconfigured to protect the speaker from excess excursion by performingoperations comprising: receiving an audio signal; applying an adaptiveexcursion protection filter, the adaptive excursion protection filteradapting filter coefficients in real-time to one or more speakerconditions based at least in part on a motion of a cone of the speakeras a function of applied voltage in the audio signal; clipping the audiosignal; applying an inverse excursion protection filter to generate theprocessed audio signal; and providing the processed audio signal to theamplifier for output to the speaker.
 12. The system of claim 11, whereinthe logic device is further configured to perform operations comprising:estimating the filter coefficients for the adaptive excursion protectionfilter and the inverse excursion protection filter based at least inpart on the amplified audio signal.
 13. The system of claim 12, whereinestimating the filter coefficients further comprises receiving currentand voltage data from the amplifier; converting the current and voltagedata to frequency domain subbands; and adaptively estimating a compleximpedance of the speaker.
 14. The system of claim 11, further comprisinga cone measurement sensor configured to physically measure speakerexcursion of the processed audio signal; and wherein the one or morespeaker conditions comprises a speaker excursion measurement.
 15. Thesystem of claim 11, wherein the logic device is further configured toperform operations comprising estimating a threshold value and applyingthe threshold value when clipping the audio signal.
 16. The system ofclaim 12, wherein estimating the filter coefficients for the adaptiveexcursion protection filter and the inverse excursion protection filterfurther comprises: estimating a complex impedance of the speaker; andestimating a direct current resistance.
 17. The system of claim 16,wherein estimating the filter coefficients for the adaptive excursionprotection filter and the inverse excursion protection filter furthercomprises: computing a motion sensitivity function representing motionof the cone as a function of applied voltage; and estimating two or morepeaks of the computed motion sensitivity function.
 18. The system ofclaim 17, wherein estimating the filter coefficients for the adaptiveexcursion protection filter and the inverse excursion protection filterfurther comprises: computing the motion sensitivity function based atleast in part on a measured DC resistance of a voice coil and pinknoise; imposing a slew-rate limit on the peak frequencies andamplitudes; and creating a model infinite impulse response filter basedat least in part on the estimated peaks.
 19. The system of claim 16,wherein the logic device is further configured to perform operationscomprising testing parameter values for a current frame of the audiosignal, the parameter values comprising one or more parameters used inthe estimation of the filter coefficients for the adaptive excursionprotection filter and the inverse excursion protection filter andidentifying the frame as invalid if a parameter is detected outside apredetermined range.
 20. The system of claim 11, wherein the logicdevice is further configured to perform operations comprising tuning theadaptive excursion protection filter through a process comprising:measuring a scale factor; determining a clipping level threshold value;and verifying the clipping level threshold value.